Best Practices Guide for Deploying SpectraLink e340, h340 and i640 Wireless Telephones

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1 Deployment Guide Best Practices Guide for Deploying SpectraLink e340, h340 and i640 Wireless Telephones Version H

2 Table of Contents 1 Introduction SpectraLink e340/h340/i640 Wireless Telephones SpectraLink Infrastructure VIEW Certification Program Wireless LAN Layout Considerations Coverage Overlapping Coverage Signal Strength Access Point Configuration Considerations Channel Selection AP Transmission Power and Capacity Interference Multipath and Signal Distortion Site Surveys Wireless Telephone Capacity Access Point Bandwidth Considerations Push-to-Talk Multicasting Considerations Telephone Usage Telephony Gateway Capacity Network Infrastructure Considerations Physical Connections Assigning IP Addresses Software Updates Using TFTP SpectraLink Voice Priority (SVP) SVP Infrastructure SVP Server Capacity Multiple SVP Servers Scenario One Scenario Two DSCP for SVP Deployments Security Wired Equivalent Privacy (WEP) Wi-Fi Protected Access (WPA) Personal, WPA2 Personal Cisco Fast Secure Roaming (FSR) Using Virtual LANs MAC Filtering and Authentication Firewalls and Traffic Filtering Virtual Private Networks (VPNs) Diagnostic Tools Subnets, Network Performance and DHCP Subnets and Telephony Gateway Interfaces Subnets and IP Telephony Server Interfaces Network Performance Requirements DHCP Requirements Conclusion

3 1 Introduction Wi-Fi telephony, also known as Voice over Wireless LAN (VoWLAN), delivers the capabilities and functionality of the enterprise telephone system in a mobile handset. The Wi-Fi handset is a WLAN client device, sharing the same wireless network as laptops and PDAs. For enterprise use, the handset is functionally equivalent to a wired desk phone, giving end-users all the features they are used to having in a wired office telephone. The benefits of VoWLAN can result in substantial cost savings over other wireless technologies by leveraging the Wi-Fi infrastructure and by eliminating recurring charges associated with the use of public cellular networks. For end users, VoWLAN can significantly improve employee mobility, resulting in increased responsiveness and productivity. Delivering enterprise-grade VoWLAN means that wireless networks must be designed to provide the highest audio quality throughout the facility. Because voice and data applications have different attributes and performance requirements, thoughtful WLAN deployment planning is a must. A Wi-Fi handset requires a continuous, reliable connection as a user moves throughout the coverage area. In addition, voice applications have a low tolerance for network errors and delays. Whereas data applications are able to accept frequent packet delays and retransmissions without the user being aware, voice quality will deteriorate with just a few hundred milliseconds of delay or a very small percentage of lost packets. Whereas data applications are typically bursty in terms of bandwidth utilization, voice conversations use a consistent and a relatively small amount of network bandwidth. Using a Wi-Fi network for voice is not complex, but there are some aspects that must be considered. A critical objective in deploying enterprise-grade Wi-Fi telephony is to maintain similar voice quality, reliability and functionality as is expected from a wired telephone. Some key issues in deploying Wi-Fi telephony include WLAN coverage, capacity, quality of service (QoS) and security. Polycom pioneered the use of VoWLAN in a wide variety of applications and environments, making the SpectraLink Wireless Telephone the market leader in this category. Based on our experience with enterprise-grade deployments, this guide provides recommendations for ensuring that a network environment is optimized for use with SpectraLink e340/h340/i640 Wireless Telephones. 1.1 SpectraLink e340/h340/i640 Wireless Telephones The information contained in this guide applies only to SpectraLink e340/h340/i640 Wireless Telephones (generically referred to as handsets throughout this document) and their OEM derivatives. Detailed product information for the SpectraLink e340/h340/i640 can be found at Polycom s web site. For information on other Polycom Wi-Fi handsets, including the SpectraLink 8020/8030 or 8002 Wireless Telephones, visit the appropriate product page at SpectraLink Infrastructure Throughout this guide references are made to SpectraLink infrastructure equipment including the SVP Server, Telephony Gateway and OAI Gateway. These LAN-based devices are sold by Polycom for use with the SpectraLink e340/h340/i640 Wireless Telephone: An SVP Server is required, as it provides the necessary WLAN QoS for the handset. Telephony Gateways allow the handset to operate as an extension off of a PBX. For systems with four or fewer Telephony Gateways, the integrated SVP Server capability can be used and a separate SVP Server is not required. For systems with more than four Telephony Gateways, a separate SVP Server is required. The OAI Gateway enables third-party applications to send and respond to real-time text messages and alerts using SpectraLink handsets. For additional details on any of these products visit the Polycom web site. 3

4 1.3 VIEW Certification Program The VIEW Certification Program is a partner program designed to ensure interoperability and maximum performance for enterprise-grade Wi-Fi infrastructure products that support Polycom s SpectraLink e340/h340/i640 and 8020/8030 Wireless Telephones and their OEM derivatives. The Program is open to manufacturers of Wi-Fi infrastructure products that incorporate the requirements described in the VIEW Technical Specification and pass VIEW Certification testing. VIEW certification requirements focus on implementing industry standards for Wi-Fi networks along with meeting the specific quality of service (QoS) and performance characteristics that are necessary for supporting Polycom handsets. For each certified product, Polycom provides a VIEW Configuration Guide that details the tested hardware models and software versions; radio modes and expected calls per AP; and specific AP configuration steps. VIEW Configuration Guides are available on the Polycom website and should be followed closely to ensure a proper deployment. 4

5 2 Wireless LAN Layout Considerations SpectraLink handsets utilize a Wi-Fi network consisting of WLAN access points (APs) distributed throughout a building or campus. The required number and placement of APs in a given environment is driven by multiple factors, including intended coverage area, system capacity, access point type, power output, physical environment, and radio types. 2.1 Coverage One of the most critical considerations in deployment of SpectraLink handsets is to ensure sufficient wireless signaling coverage. Enterprise Wi-Fi networks are often initially laid out for data applications and may not provide adequate coverage for voice users. Such networks may be designed to only cover areas where data devices are commonly used, and may not include coverage in other areas such as stairwells, break rooms or building entrances all places where telephone conversations are likely to occur. The overall quality of coverage is more important for telephony applications. Coverage that may be suitable for data applications may not be seamless enough to support the requirements of VoWLAN. Most data communication protocols provide a mechanism for retransmission of lost or corrupted packets. Delays caused by retransmissions are not harmful, or even discernable, for most data applications. Likewise, data applications will typically tolerate more retries than voice applications, as the useful life of a voice packet is short. However, the real-time nature of a full-duplex telephone conversation requires that voice packets be received correctly within tens of milliseconds of their transmission. There is little time for retransmission, and lost or corrupted packets must be discarded after limited retries. In areas of poor wireless coverage, the performance of data applications may be acceptable due to retransmission of data packets, but for real-time voice, audio quality will likely suffer. Another factor to consider when determining the coverage area is the device usage. Wireless telephones are used differently than wireless data devices. Handset users tend to walk as they talk, while data users are usually stationary or periodically nomadic. Wireless voice requires full mobility while data generally requires simple portability. Wireless handsets are typically held close to the user s body, introducing additional radio signal attenuation. Data devices are usually set on a surface or held away from the body. The usage factor may result in reduced range for a wireless telephone as compared with a data device. Therefore, the WLAN layout should account for some reduction of radio signal propagation Overlapping Coverage Wi-Fi cell overlap must be considered when planning your VoWLAN deployment. Handsets make a determination to roam in less than half the overlapping coverage area. Therefore, the coverage area must be adequate enough so that when a voice user is moving, the handset has time to discover the next AP before signal on the existing AP becomes too weak. A properly designed Wi-Fi network will position APs with sufficient overlapping coverage to ensure there are no coverage gaps, or dead spots, between them. The result is seamless handoff between APs and excellent voice quality throughout the facility. Sufficient overlapping coverage is usually considered 15% to 20% signal overlap between AP cells in a deployment utilizing maximum transmit power for both handsets and APs. Smaller cells will need larger overlaps due to the potential for much smaller cell size which causes a decrease in overall overlap from a maximum transmit power deployment. The 15% to 20% of signal overlap between AP cells generally works well with a typical walking speed of the user (the average walking speed of an individual is 3 mph). If the speed of the moving user is greater (such as a golf cart, fork lift or running/jogging) then a different overlap strategy may be necessary for successful handoff between APs. The WLAN layout must factor in the transmission settings that are configured within the APs. The transmission of voice requires relatively low data rates and a small amount of bandwidth compared to other applications. The standard includes automatic rate switching capabilities so that as a user moves away from the AP, the radio adapts and uses a less complex and slower transmission scheme to send the data. The result is increased range when operating at reduced transmission data rates. When voice is an application on the WLAN, APs should be configured 5

6 to allow lower transmission rates in order to maximize coverage area. If a site requires configuring the APs to only negotiate at the higher rates, the layout of the WLAN must account for the reduced coverage and additional APs will be required to ensure seamless overlapping coverage. SpectraLink handsets perform Dynamic Channel Assessment (DCA) in between the transmission of packets to learn about neighboring APs. It takes about one second for a DCA cycle to complete for a standard three channel deployment for b. In order to ensure a DCA cycle can complete within the assessment area (see Figure 1), a person moving through the assessment area must be within the area for at least 4-5 seconds to make sure the DCA starts and ends within the assessment area. Failure to complete the DCA cycle within the assessment area can lead to lost network connectivity resulting in a hard handoff, lost audio, choppy audio or potentially a dropped call. Figure 1 - Dynamic Channel Assessment (DCA) The handset compares the signal strength of neighboring APs to determine whether to roam from the current AP. In order to roam, the handset has to determine whether other APs are either five decibels (db) (for any first attempt associating with an AP) or ten decibels stronger (to roam back to the previous AP) than the current AP s signal. In most cases the handset only needs five decibels of signal difference between APs to make a decision to roam. But to prevent ping-pong behavior the separation needs to be ten decibels higher for the handset to return to the previously associated AP. This behavior requires that the assessment area must have at least a ten decibel difference to enable good roaming behavior for all cases. Corners and doorways pose a particular design issue. The shadowing of corners can cause steep drop-offs in signal coverage. Make sure to have adequate cell overlap at and around corners so that the audio stream is not impacted by a user going around corners. This may require placement of AP at corner locations to ensure appropriate cover and prevent RF shadows. 6

7 2.1.2 Signal Strength To provide reliable service, wireless networks should be engineered to deliver adequate signal strength in all areas where the wireless telephones will be used. The required minimum signal strength for all SpectraLink handsets depends on the data rates enabled on the AP and may also require consideration for the frequency band and modulation used. Recommended signal strength characteristics are summarized in Table 1 & Table 2, below. Use these values to determine the outer edge of an AP cell boundary as shown, in Figure 1 above, as the limit of AP A or limit of AP B. The value to use is dependent on the highest WLAN data rate set Mandatory. Typically, most WLAN APs transmit beacons and multicast frames at the highest rate set Mandatory Rate (Mb/s) Best Practices (dbm) 2.4GHz b (CCK) Table 1 2.4GHz The critical factor is the highest data rate set to Required or Mandatory 1. Other data rates can be set to Supported. The highest data rate set Mandatory determines the RF power output required by the wireless telephone for proper operation. Broadcast frames (beacons) utilize the highest Basic 2 data rate and multicast frames (used for the SpectraLink i640 s push-to-talk feature and SRP handset check-ins) also use the highest data rate set Mandatory. Unicast frames (data) use the best or highest data rate of all available rates which supports low retries and low packet errors. As errors and retries increase the unicast data rate will scale down to lower available data rates. Referencing Table 1 the highest rate set Mandatory (Required) determines the signaling requirements for the wireless telephone in all areas where they are used. For example, if an b/g access point has 1Mbps, 2Mbps, 5.5Mbps and 11Mbps all set Mandatory, the handset requires -65dBm in all areas. For example, if an b/g access point has 1Mbps Mandatory and other rates set Supported (or Enabled ) the handset requires -75dBm in all areas. SpectraLink handsets have a Site Survey mode that can be used to validate the signal strength it is receiving from the AP. The handset also has a Diagnostics mode which can show AP signal strength, as well as other details, as received during a call. See the SpectraLink e340/h340/i640 Wireless Telephone Administration Guide for details on using the Site Survey and Diagnostics mode features. Although it is possible that SpectraLink handsets may operate at signal strengths which are weaker than those provided in Table 1, real world deployments involve many RF propagation challenges such as physical obstructions, interference, and multipath effects that impact both signal strength and quality. Designing RF coverage to the 1 Access Point (AP) vendors refer to this configuration setting differently but the value indicates a data rate that clients must be capable of utilizing in order to associate with the access point. These data rates are also used for different data traffic types by clients and APs that should be considered when designing for coverage requirements. 2 The Standard defines any data rate set as required to be basic rates. See for additional details. ( 7

8 required levels will provide an adequate buffer for these propagation challenges, enabling a more reliable and consistent level of performance with low retry rates. 2.2 Access Point Configuration Considerations There are several fundamental access point configuration options that must be considered prior to performing a site survey and deploying a voice-capable WLAN infrastructure. In general, adjacent APs in three dimensions (above, below and beside) must use different non-overlapping radio channels to prevent interference between them. This document does not cover all issues or considerations for WLAN deployment. It is strongly recommended that Polycom Professional Service, or another suitable professional services organization, with wireless voice deployment experience be engaged to answer additional questions about configurations that may affect voice quality or wireless telephone performance. In addition, VIEW Configuration Guides for WLAN infrastructure, which are available from the Polycom web site, should be followed closely Channel Selection The b standard provides for three non-interfering, non-overlapping frequency channels - channels one, six and eleven in North America. Access points within range of each other should always be set to non-interfering channels to maximize the capacity and performance of the wireless infrastructure. Figure 2 illustrates the correct deployment methodology for b deployments. Figure b Non-interfering Channels with Overlapping Cell Coverage If adjacent access points in three dimensions (above, below or beside) are set to the same channel, or utilize channels with overlapping frequency bands, the resulting interference will cause a significant reduction in the network performance and throughput, and will degrade overall voice quality. A space of twenty five MHz, or 5 channels or greater should be used to configure neighbor APs for non-interfering channels Figure 3 represents the 2.4 GHz frequency range, indicating the overlap in channel frequencies. 8

9 MHz MHz MHz Figure b Channels AP Transmission Power and Capacity The AP transmit power should be set so that the handsets receive the required minimum signal strength, as defined in Section of this document. For deployments with higher AP density, lower transmit power settings are typically required to prevent channel interference. Maximum AP power settings vary by band and by channel, and can vary between countries. Local regulations should always be checked for regulatory compliance considerations. In addition, maximum power output levels may vary by AP manufacturer. Where possible, all APs should be set to the same transmit power level within a given radio type. It is crucial to then set the transmit power of the handset to match the transmit power of the APs for that band. This will ensure a symmetrical communication link. Mismatched transmit power outputs will result in reduced range, poor handoff, one-way audio and other quality of service or packet delivery issues. SpectraLink Wireless Telephones support transmission power settings in the range from 5mW to 100mW (in the United States). The transmit power setting on the handset should match the AP s actual configured transmit power. Any AP antenna gain will increase signal gain in both directions. Regardless of the selected power level settings, all APs and handsets must be configured with the same settings to avoid channel conflicts or unwanted cross-channel interference. For access points that support automatic transmission power adjustments, Polycom recommends using only static power settings to ensure optimal performance. In mixed b/g environments, Polycom recommends configuring the transmit power of the b and g radios to the same setting, if they are separately configurable. For example, set both radios to 30mW to ensure identical coverage on both radios. For mixed a/b/g environments, where the AP utilizes all three radios types, AP placement should first be determined by modeling for the characteristics of a, since this environment will typically have the shortest range. Then, the transmit power of the b and g radios should be adjusted to provide the required coverage levels and cell overlap for those networks, within the already established AP locations Interference Interference on a wireless network may originate from many sources. Microwave ovens, Bluetooth devices, cordless phones, wireless video cameras, wireless motion detectors, and rogue APs are among the many potential interfering RF (radio frequency) sources. In general, devices that employ or emit radio frequency signals within a given radio coverage area will have the potential to cause unwanted interference. 9

10 Radio frequency spectrum analyzers can be used to help identify the sources of such interference. Once identified, interference is best mitigated by removing the interfering device(s) from the network area. Otherwise, it may be possible to change the channel setting of the interfering device to avoid conflict with the surrounding APs. If this is also not possible, then it may be possible to change the channel of the surrounding APs to avoid as much radio frequency overlap with the interfering device. A documented facility-wide radio frequency usage policy will help control sources of RF energy. Ideally, any RF generating device should have prior approval before introduction onto the property or installation in any building or structures Multipath and Signal Distortion Multipath distortion is a form of RF interference that occurs when a radio signal has more than one path between the transmitter and the receiver causing multiple signals to be detected by the receiver. This is typically caused by the radio signal reflecting off physical barriers such as metal walls, ceilings and other structures and is a very common problem in factories and storage environments. Multiple converging wave fronts may be received as either an attenuated or amplified signal by the receiver. In some instances, if the signals arrive exactly out of phase, the result is a complete cancellation of any RF signal. Multipath can cause severe network throughput degradation because of high error rates and packet retries. This in turn can lead to severe voice quality impairment with SpectraLink Wireless Telephones. Correctly locating antennas and choosing the right type of antenna can help reduce the effects of multipath interference. AP diversity antennas should always be used to help improve performance in a multipath environment. A diversity solution uses two antennas for each radio, and will send and receive signals on the antenna which is receiving the best signal from the wireless client. Diversity in an AP with two antennas, which provide signaling to the same geographic area, provides a unique signal path from each antenna to the handset. This greatly increases the probability that both the AP and the handset will receive a better signal quality in multipath environments. Most access points support receive diversity in that they accept the received transmission on the antenna that is getting the best signal. Some also support full transmit diversity where the transmission is made on the same antenna that was last used to receive a signal from that specific client. In order to provide optimal voice quality, Polycom recommends the use of APs supporting both receive and full transmit diversity in environments where multipath is an issue. This will help optimize the WLAN for all wireless clients. External antennas provide additional flexibility in type (omni or directional), mounting options and gain. External antennas can be separated from 4.5 inches to 5 feet at each AP radio. Access point antennas should not be placed near a metal roof, wall, beam or other metal obstruction in any environment, as this will amplify the reflection effects. Additionally, antennas should be positioned so that they have line of sight (LoS) to most of the clients that they service. Additional instructions from the wireless network infrastructure vendor should be followed with regard to antenna selection and placement to provide correct diversity operation Site Surveys A wireless RF site survey is highly recommended for any wireless network deployment. However, it is especially critical for VoWLAN and is essential for large or complex facilities. An RF site survey can ensure that the wireless network is optimally designed and configured to support voice by confirming RF placement, cell overlap, channel allocation/reuse, packet transmission quality, packet retry rates, and other deployment considerations. While many tools exist that allow customers to perform their own assessment, Polycom recommends a professional site survey to ensure optimum coverage and minimum interference. Polycom offers a full suite of site-survey services that will ensure a WLAN is properly configured to support wireless voice. To verify coverage of an installed Wi-Fi network, Polycom handsets offer a site-survey mode that can be used to validate the AP locations and configurations are both correct and adequate. This mode detects the four strongest AP 10

11 signals and displays the signal strength along with the AP channel assignments. The site survey mode may be used to detect areas with poor coverage or interfering channels; check for rogue APs; confirm the Service Set Identification (SSID) and data rates of each AP and include the security and QoS mechanisms supported by the AP; and detect some AP configuration problems. With SpectraLink handsets, the entire coverage area must be checked to ensure that at least one access point s output meets the signal strength requirements summarized in Section of this document. If the site-survey mode indicates that two APs are using the same channel within range of the handset, it is important to adjust the channelization to avoid channel conflicts. After a site survey is complete, coverage issues can be resolved by adding and/or relocating APs if necessary. Overlap issues may be resolved by reassigning channels or by relocating some access points. When adjustments are made to the WLAN configuration an additional site survey or site verification should be performed to ensure that the changes are satisfactory and have not had an adverse impact in other areas of coverage. 2.3 Wireless Telephone Capacity Network capacity requirements factor into the number of APs required, although in most cases the coverage area is the primary factor. Data traffic is often very bursty and sporadic. This is typically acceptable because data applications can tolerate network congestion with reduced throughput and slower response times. Voice traffic cannot tolerate unpredictable delays, where the bandwidth requirements are much more constant and consistent. Voice traffic can also be predicted using probabilistic usage models, allowing a network to be designed with high confidence in meeting anticipated voice capacity requirements. Beyond the standard IP telephony design guidelines, there are several additional considerations that should be addressed for VoWLAN with SpectraLink handsets. The SVP Server prevents oversubscription of an AP and improved load balancing by limiting the maximum number of active calls per AP. Maximum settings, using the highest data rate, are AP specific and can be found in the VIEW Configuration Guides on the Polycom web site. The SVP Server value determines the maximum number of wireless telephones in-call on a given AP and forces handsets to handoff (roam) when capacity maximums are reached. Overall, the calls per AP specified in the SVP Server is often lower than the maximum number an individual AP may be able to support. This allows some telephones to work at lower rates (1Mbps and 2Mbps) and some at the highest data rates Access Point Bandwidth Considerations There are several factors which determine the AP bandwidth utilization during a telephone call. The first is the VoIP protocol used and its characteristics. The type of codec utilized combined with the packet rate will determine the size of the voice packets along with any additional overhead information required for the protocol. Payload data will generally account for 30-50%of a typical voice packet, with and IP protocol overhead filling the rest. The protocols include timing gaps for collision avoidance, which means bandwidth utilization is more accurately quantified as a percentage of available throughput rather than actual data throughput. The percentage of bandwidth required is greater for lower b data rates; however it is not a linear function because of the bandwidth consumed by the timing gaps and overhead. For example, a call using standard 64 Kbps voice encoding (G.711) utilizes about 4.5 percent of the AP bandwidth at 11 Mbps, and about 12 percent at 2 Mbps. In this example, four simultaneous calls on an AP would consume about 18 percent of the available bandwidth at 11 Mbps or about 48 percent at 2 Mbps or about 90 percent at 1Mbps. The maximum number of simultaneous telephone calls an AP can support is determined by dividing the maximum recommended bandwidth usage by the percentage of bandwidth used for each individual call. Note that approximately 20 to 35 percent of the AP bandwidth must be reserved for channel negotiation and association algorithms, occasional retries, and the possibility of occasional transmission rate reductions caused by interference or other factors. Therefore, 65 to 80 percent of the total available bandwidth should be used for calculating the maximum call capacity per AP. For example, if all calls on an AP are using a theoretical 5.4 percent of the bandwidth at 11 Mbps, the actual number of calls expected at that rate would be about 12 (65 percent of bandwidth available / 5.4 percent theoretical bandwidth utilized per call). Lower overall bandwidth is available when there are a greater number of devices associated with an AP or when lower data rates are used for the telephone call or calls. 11

12 Even with all of the known variables, there are many other vendor-specific characteristics associated with individual APs that make it difficult to quantify the precise number of concurrent calls per AP, without thorough testing of specific configurations. Polycom s VIEW Configuration Guides identify the maximum number of calls per AP for specific models that have been tested to be compatible with the SpectraLink handset. With SVP, Polycom provides the ability to limit the number of calls per AP with a configurable setting in the SVP Server. The Calls per Access Point setting limits the number of active calls on each AP and can be used to set aside bandwidth for data traffic. Wireless Telephones in-call are free to associate with other APs within range that have not reached the set maximum number of calls. Polycom requires this setting to be equal to or below the maximum number of calls specified in VIEW Configuration Guides. It is still possible for the number of phones associated to an AP to exceed the maximum number of calls as would be the case with any additional clients associating to the same AP. The maximum number of calls per AP will simply control how many of those associated phones will be able to enter into a call. Additional handsets beyond the maximum number specified will be forced by the SVP Server to roam to a new AP that has not reached the maximum calls per AP. If no APs are available any handsets beyond the maximum will display an error message indicating there is insufficient bandwidth to complete the call Push-to-Talk Multicasting Considerations SpectraLink i640 handsets provide push-to-talk (PTT) functionality using the Polycom-proprietary SpectraLink Radio Protocol (SRP) ADPCM encoding. Because the PTT mode uses IP multicasting, all APs on the subnet will transmit a PTT broadcast. This can be limited to only the APs that are handling one or more PTT-enabled handsets by enabling the Internet Group Management Protocol (IGMP) on the wired infrastructure network. Ensure the LAN is configured to propagate PTT (multicast) traffic to all WLAN APs used by the SpectraLink handsets. When i640 handsets are deployed on a network with newer versions of SpectraLink handsets, some interoperability considerations must be observed. The newer SpectraLink 8030 handsets have 24 PTT channels plus one priority channel available. SpectraLink i640 handsets have eight PTT channels with no priority channel. When PTT is activated on a network using a mix of handset versions, only the eight common channels will be available for the i640 handsets Telephone Usage When the handset is used with traditional PBXs through a Telephony Gateway, the PBX interface will assemble audio, packetize it, and release these packets at a preset interval. The PBX release interval is generally 20ms or 30ms. The SVP Server will receive these audio packets and release them to the network for delivery to the handset every 30ms. With a PBX release interval of 20ms, packets delivered to the handset by the SVP Server will have one audio payload followed by a packet with two audio payloads. This pattern, one audio payload then two audio payloads, will continue during the call. With a PBX release interval of 30ms, packets sent to the handset will have one audio payload each. In rare occasions, a PBX may use a 40ms release interval. With this audio payload release interval, packets delivered to the wireless telephone will have one large audio payload or no audio payload per packet sent to the handset. The no audio payload packets and long time between audio (two SVP packets 60ms) payload aggravates any weakness (multi-path, retry packets, etc.) in the WLAN and will cause poor audio. Therefore, whenever possible the PBX should be configured to use release intervals of 30ms or 20ms. Because data rate and packet rates are constant with voice applications, wireless telephone calls may be modeled in a manner very similar to circuit-switched calls. Telephone users (whether wired or wireless) generally tend to make calls at random times and of random durations. Because of this, mathematical models can be applied to calculate the probability of calls being blocked based on the number of call resources available. Telephone usage is measured in units of Erlangs. One Erlang is equivalent to the traffic generated by a single telephone call in continuous use. A typical office telephone user will generate 0.10 to 0.15 Erlangs of usage during normal work hours, which equates to six to nine minutes on the telephone during an average one-hour period. Heavy telephone users may generate 0.20 to 0.30 Erlangs, or an average of 12 to 18 minutes of phone usage in an hour. 12

13 Note that traffic analysis is based on the aggregate traffic for all users, so users with higher or lower usage are included in these averages. The traffic engineering decisions are a tradeoff between additional call resources and an increased probability of call blocking. Call blocking is the failure of calls due to an insufficient number of call resources being available. Typical systems are designed to a blocking level (or grade of service) of 0.5 percent to two percent at the busiest times. Traffic model equations use the aggregate traffic load, number of users and number of call resources to determine the blocking probability. The blocking probability can also be used along with the aggregate traffic load to determine the number of call resources required. Traffic model equations and calculators are available at Consider a system with APs that can support six active telephone calls. If a blocking probability of one percent or less is desired, each AP can support approximately 13 moderate wireless telephones users. If the AP coverage supports 12 simultaneous calls per AP, each AP can then support approximately 39 moderate users. This allows some users to be in-call and others in standby. The Table 2 shows maximum users per AP based on the AP s ability to handle simultaneous calls: User Calling Intensity Light Moderate Heavy Erlangs per User Max Active Calls per AP Users Supported per AP (1% Blocking Probability) Table 2 - Users Supported per Access Point Areas where heavier wireless telephone usage is expected, such as cafeterias, staff lounges, and auditoriums, can obtain higher call capacity and handle more users by installing additional APs. For most enterprise applications however, the table above should be sufficient in demonstrating the number of wireless handsets supported within each AP s coverage area Telephony Gateway Capacity Telephone system administrators should consider the user distribution on SpectraLink 8000 Telephony Gateways much in the same way as they do PBX line cards. Telephony Gateways incorporate a physical connection to a PBX line card. The phone system administrator should spread departments or functional areas across multiple PBX line cards and across multiple Telephony Gateways so that a failure of either component does not cause a complete wireless handset outage in one department or area. In addition, system administrators must consider that one Telephony Gateway can support a maximum of eight handsets in an active call state. While the Telephony Gateway 13

14 can manage 16 wireless telephones total only eight can be in call at any one time. Therefore, heavy users should be spread across Telephony Gateways to reduce the chance of call blocking. 14

15 3 Network Infrastructure Considerations 3.1 Physical Connections SpectraLink infrastructure components, including the SVP Server(s), Telephony Gateways and OAI Gateway, must connect to a facility s LAN using enterprise-grade Ethernet switches rather than Ethernet hubs or consumer-grade SOHO switches in order to provide adequate bandwidth and limit traffic collisions and bottlenecks (see Figure 4 for reference). Ethernet switches should be configured to statically set the speed and duplex values as appropriate for the device being connected to that port. The SVP Server should be set to the 100Base-T/Full Full-duplex transmission setting. This is required to support the maximum simultaneous voice calls and for optimal system performance. The SpectraLink Telephony Gateway and OAI Gateway products utilize a 10Base-T, half-duplex Ethernet interface and the Ethernet switch ports should be set accordingly. Network wiring is an important component of any Ethernet-based system and is subject to local and state building code specifications. Cat 5 or better, 4-pair 10/100 Base-T Ethernet cabling must be used for SpectraLink infrastructure equipment. Wireless bridges are sometimes used to interconnect geographically isolated Ethernet LANs or to extend the range of existing WLANs. Such devices create bottlenecks for network capacity and add delay to the overall network, which are generally not tolerable for real-time voice connections. Polycom does not support a configuration that includes wireless bridges and does not recommend using wireless bridges with any wireless network supporting voice. Figure 4 Physical Connections 15

16 3.2 Assigning IP Addresses SpectraLink handsets operate as LAN client devices and therefore require IP addresses to operate in the network. IP addresses can be assigned statically through the configuration menus on the handsets or dynamically using standard DHCP protocol. The Handset Administration Tool (HAT) can be used to quickly load and change administration options in the handsets, including static IP addresses. For dynamic IP addressing, a DHCP server is required. Telephony Gateways and SVP Servers also require IP addresses that can be obtained by either static or DHCP address assignment. It is always recommended to configure production infrastructure components with static IP addresses to ensure consistent system access. When using one or more SVP Server(s), (see Section 4.1.3) the Registration SVP Server must be assigned a static IP address. The Registration SVP Server is identified by DHCP option 151 to the wireless telephones. When operating with an IP telephony server (IP PBX), other than Avaya or Cisco, the SVP Server also requires a range of IP addresses that cover the total number of wireless telephones supported by that SVP Server. That range of IP addresses is known as First Alias IP Address/Last Alias IP Address in the SVP Server configuration menu. It is important to note that for redundancy purposes it may be necessary to assign more IP addresses to an SVP s Alias IP range than what the SVP Server would normally support. Each SVP Server supports up to 500 handsets registered, but this can be limited by the total number of Alias IP addresses configured in the SVP Server. When a handset is using SVP and registers with the telephony server, one of the IP (Alias) addresses within this range is used to communicate between the SVP Server and the telephony server. This IP address is used by the SVP Server as an alias to communicate with the telephony server on the wireless telephone s behalf, but will not be equivalent to the handset s IP address that was either statically assigned or obtained from the DHCP server. The range of alias IP addresses must not be used within any DHCP range or cover the IP address used by any other device. In the case where multiple SVP Servers are used for added capacity or redundancy, an exclusive range of IP addresses equivalent to the number of total users each SVP Server supports is required per SVP Server. All alias IP addresses must be within the same IP subnet as the IP address of the SVP Server they are assigned to. 3.3 Software Updates Using TFTP All SpectraLink infrastructure components are field-upgradeable in terms of new software features and bug fixes. SpectraLink handsets utilize a TFTP client to automatically download new code when available. Deployments using Telephony Gateways to connect to a traditional PBX have an integrated TFTP server to support Wireless Telephone and OAI Gateway software upgrades. While, the integrated TFTP server can be used to deliver software to the e340/h340/i640 handsets, the Telephony Gateway cannot hold both handset type (e340/h340/i640 and 8020/8030) software files due to memory allocation. A network TFTP server will simultaneously update multiple handsets, while the Telephony Gateway can only update handsets one at a time. Therefore, in larger systems and newer deployments, a separate TFTP server should be used rather than using the Telephony Gateway s TFTP server capability. For deployments with multiple Telephony Gateways it is recommended to utilize an external TFTP server to centralize the management and delivery of software. The SVP Server also requires a TFTP server for software updates. The Telephony Gateway cannot be used as a TFTP server for the SVP Server code. Telephony Gateways receive software updates only through manual FTP updates. The OAI Gateways can receive software updates via FTP as well but if software recovery becomes necessary the OAI will utilize a TFTP server. The latest software versions are available from Polycom s web site. 16

17 4 SpectraLink Voice Priority (SVP) Polycom pioneered VoWLAN for the enterprise and remains the market leader today. One key success factor has been our SpectraLink Voice Priority (SVP) mechanism for QoS. This method is proven to deliver enterprise-grade voice quality, battery life and call capacity for SpectraLink handsets. Quality of service (QoS) is a means of providing a level of service that will result in a network connection of acceptable quality. Typically this results in providing different levels of service for different applications, depending on their requirements. When data and voice are competing for bandwidth, such as in a WLAN, it is necessary to have mechanisms to prioritize voice packets over data, preserve battery life for handhelds, and allocate appropriate AP bandwidth for the associated device s applications. The original standard did not provide a QoS mechanism, so Polycom developed SVP to allow delay-sensitive voice and asynchronous data applications to coexist on a Wi-Fi network without compromising voice quality. Excellent voice quality for SpectraLink handsets is ensured on a shared Wi-Fi network using SVP. Adopted by the majority of enterprise-class WLAN vendors, SVP is well-proven and guarantees audio quality on a shared voice and data network. SVP is compatible with standards, but uses proprietary methods for packet prioritization, battery management and call admission control. Access points generally use random back-off intervals that require all types of traffic to contend for access to the wireless medium with equal rights. However, treating all traffic equally can cause significant delays to voice traffic. Modifying the AP behavior to recognize and prioritize voice packets increases the probability of better performance while continuing to treat asynchronous data packets normally. The two operations that comprise SVP in the AP, minimizing random back-off and priority queuing, require a packetfiltering mechanism. Packet filtering requires recognizing the packet s type. SpectraLink packets are registered as IP protocol ID 119 at layer 4. The SVP Server performs packet delivery timing through the AP to the wireless telephones, which is critical for ensuring seamless handoffs among APs and for enhanced battery management. The following section offers a more detailed explanation of timed delivery SVP Infrastructure To trigger SVP in the APs from the wired side of the network, a Telephony Gateway with integrated SVP Server and/or a standalone SVP Server is required. Telephony Gateways can provide SVP support for small installations with four or fewer Gateways. A SVP Server is required for applications using an IP telephony server or using more than four Telephony Gateways SVP Server Capacity A single SVP Server supports 120 simultaneous calls when used with Telephony Gateways or 80 simultaneous calls with an IP telephony server. Multiple SVP Servers can be used to increase capacity to support up to 850 total calls (which can support approximately 8,000 Wireless Telephones) for IP telephony server interfaces. When used with Telephony Gateways, the total number of users is limited to 640 (40 Telephony Gateways). For smaller IP telephony interface deployments, 10 and 20-user SVP Servers are available. Refer to Polycom s SpectraLink 8000 SVP Server Administration Guide for additional information regarding the maximum number of simultaneous calls and wireless telephones supported by multiple SVP Servers Multiple SVP Servers For installations with multiple SVP Servers, call resources are automatically allocated between the APs and the SpectraLink Wireless Telephones by those devices Media Access Control (MAC) addresses. In most instances, because of the large number of wireless telephones and APs expected in such an application, the distribution of call processing will be relatively even across all SVP Servers. Some installations with multiple SVP Servers (SVP code < 17x.033) are configured to have primary ( master ) and one or more secondary ( slave ) servers. If a secondary SVP Server fails and can no longer be detected, the packet handling (and associated handsets) will automatically be redistributed among the remaining servers. All active calls associated to the failed secondary SVP server will be lost during this process, however the affected wireless telephones will check-in with available SVP servers without manual reconfiguration. In the case of a master SVP 17

18 server failure, the wireless telephone system will be disrupted. To minimize downtime related to a failed master SVP Server or a single server, it is recommended that a spare SVP Server be readily available. The network administrator can assign the IP address of the failed unit to the replacement SVP Server. Alternatively the number of SVP servers can be scaled to ensure that if one or more SVP servers fail that all handsets can be allocated to the remaining SVP servers. This will require that sufficient alias IP addresses be made available on all SVP servers to support the allocation of additional handsets to the remaining SVP Servers. More recent installations with multiple SVP Servers (SVP code 17x.033) use the SVP Self Healing feature and do not use the Master/Slave concepts of earlier versions. There is, however, a designated primary SVP Server, called the Registration SVP Server, that has its IP Address defined either statically in the Wireless Telephone network configuration or acquired from DHCP option 151, thus allowing the Wireless Telephone to initially check-in to the telephone system. Updated handset firmware is required to take full advantage of SVP Self-Healing functionality. See Table 3 for the firmware revisions where SVP Self-Healing functionality was first introduced. SVP 17x.033 with Handset model Handset code Avaya 3616/3620/ NEC MH110/120/ Nortel 2210/2211/ SIP e340/h340/i Table 3 Handset Code Versions That Support SVP Self-Healing The SVP Server acts as a proxy for the handset by sending and receiving packets to/from the call server or PBX. In some IP implementations, the SVP Server also performs Network Address Translation (NAT) for the handset. The main functions for the SVP Server to perform are indicated in Table 4. Function SVP Server 1 (Registration) SVP Server 2 Manage handsets Manage voice packet delivery by the AP Proxy between voice platform and handset Send/receives all packets to/from handset Considered home SVP Server Limit maximum handsets in-call per AP (static number entered by administrator) Proxy between voice platform and handset Send/receives all packets to/from handset Considered home SVP Server Limit maximum handsets in-call per AP (static number entered by administrator) Receive packets from the home SVP Server and forward to handset though currently associated AP Table 4 SVP Server Functions Receive packets from the home SVP Server and forward to handset though currently associated AP The process by which the handset is able to get onto the network and register with the SVP Server and telephony platform is a handshake process which requires multiple steps. Figure 5 is a reference followed diagram by the step by step description of the handset and SVP Server packet handshake. 18

19 Figure 5 Multiple SVP Servers Handset 1 registers with SVP Server 1 o Handset 1 always sends its packets to SVP 1 SVP 1 forwards handset 1 packets to the call server or PBX The call server or PBX sends packets from the telephone this handset is in-call with to SVP 1 SVP 1 sends packets to the SVP managing the handset 1 s AP o When handset 1 is using AP A (associated with) it receives packets from SVP 1 o When handset 1 is using AP B it receives packets from SVP 2 o When handset 1 is using AP C it receives packets from SVP 2 o When handset 1 is using AP D it receives packets from SVP 1 Handset 2 registers with SVP Server 1 o Handset 1 always sends its voice packets to SVP 1 SVP 1 forwards handset 2 audio packets to the call server or PBX The call server or PBX sends packets from the telephone this handset is in call with to SVP 1 SVP 1 sends packets to the SVP managing handset 2 s AP o When handset 2 is using AP A it receives packets from SVP 1 o When handset 2 is using AP B it receives packets from SVP 2 o When handset 2 is using AP C it receives packets from SVP 2 o When handset 2 is using AP D it receives packets from SVP 1 Handset 3 registers with SVP Server 2 o Handset 3 always sends its packets to SVP 2 SVP 1 forwards handset 3 audio packets to the call server or PBX The call server or PBX sends audio packets from the telephone this handset is in call with to SVP 2 SVP 2 sends packets to the SVP managing handset 3 s AP o When handset 3 is using AP A it receives packets from SVP 1 o When handset 3 is using AP B it receives packets from SVP 2 o When handset 3 is using AP C is receives packets from SVP 2 o When handset 3 is using AP D it receives packets from SVP 1 19

20 Scenario One Scenario One assumes that the handset has registered to SVP Server 1, associated to Access Point A, and managed by SVP Server 1 (Figure 6) Handset communicates only to SVP Server 1 in this case Figure 6 - Scenario One Scenario Two Scenario Two assumes that handset has roamed to Access Point B and is managed by SVP Server 2 (Figure 7) Packets to handset (from call server or PBX) are first sent through its Home SVP Server (Server 1), then forwarded to SVP Server 2, and then transmitted by AP B to the handset On the return trip the handset communicates back through AP B to its Home SVP Server (Server 1) and back to the call server or PBX Figure 7 - Scenario Two Calls between handset may bypass the call server or PBX and connect directly between SVP Alias IP addresses. Control messaging will still go to /from the call server or PBX. As a final note, the wireless telephone learns about all available SVP Servers IP addresses when it powers up and does the SRP Check-In with the Registration SVP Server (identified in DHCP option 151 or set statically). Once the 20

21 handset are aware of all SVP Servers even if the Registration SVP Server identified by DHCP option 151 goes down, the handset will still find another SVP Server to use. This is the SVP self-healing concept in action. Any new or replacement wireless telephones may fail to Check-In if the SVP Server identified by DHCP option 151 is down. The new handset has not yet learned the list of all SVP Servers and does not get a response to its Check-In Request. In this case, check the SVP Server identified by DHCP option 151 for operational status. Note that it is not possible to create an IP array of the SVP Servers in the DHCP option as the handset would not understand this and would instead display No SVP Response on the screen. DHCP option 151 is required for the handset when it is operating with an IP protocol but is not required when the handset is working in an Telephony Gateway environment. When wireless telephones do the first SRP Check-In Request with the registration SVP Server, the registration SVP will load balance the handsets across all available SVP Servers. It will tell the wireless telephone to Check-In with another SVP Server based on a load balancing algorithm. This algorithm does not guarantee an even distribution of handset across all SVP Servers. The wireless telephone will do an SRP Check-In with this designated SVP Server. Once connected to the SVP Server the handset will establish its connection to the call server or PBX, then will become operational and be ready to make or receive calls DSCP for SVP Deployments Quality of Service on the wired network must also be taken into consideration. With any VoIP system there is a need for QoS to be configured from end-to-end in order to effectively ensure good audio quality and performance. Even systems that are implemented with only voice and no data clients there is still a need for QoS to ensure background services do not interfere with the audio. The SVP Server is responsible for providing packet prioritization and timed release for specific time slot deliveries to all wireless phones. Additionally, the SVP Server must have its QoS tagging configured to use DSCP (Differentiated Services Code Point) tags that will be properly recognized by the wired and wireless network in order to provide for low latency packet transport. There are six different DSCP tags that can be configured in the SVP Server. It is best to always refer to the network manufacturer s documentation to see what DSCP tags are supported natively and which will need to be configured to give high priority to all the of handset s traffic. By default all SVP Server DSCP tags are set to a value of zero or four. A valid DSCP tag range is from 0 to 63. The SVP Server has the following packet types to set with DSCP tags: Administration Network administration services such as telnet and FTP In Call All traffic from the SVP Server to handset during an active call Standby All traffic from the SVP to handset during standby RTP All in-call audio traffic from the SVP Server to call server or PBX PBX All control traffic from the SVP Server to the call server or PBX Inter-SVPII All traffic sent to other SVP Servers The recommended values for these traffic types should be determined by your network infrastructure manufacturer and/or call server or PBX manufacturer. However, in most environments specific DSCP tag values are considered to be the expected or default DSCP values. The recommended DSCP tag values for the defined traffic types are: Administration Default or 0 In Call 46 Standby 26 (If using PTT the DSCP tag should be changed to 46) RTP 46 PBX 24 Inter-SVPII 46 21

22 All DSCP values defined here, regardless of what values are used, must be configured for appropriate priority throughout the network to ensure end-to-end QoS functionality. It is critical that voice receive the highest priority to ensure the user experience is as good as possible. 22

23 5 Security Proper security provisions are critical for any enterprise Wi-Fi network. Wireless technology does not provide any physical barrier from malicious attackers since radio waves penetrate walls and can be monitored and accessed from outside the facility. The extent of security measures used is typically proportional to the value of the information accessible on the network. The security risk for VoWLAN is not limited to the typical wired telephony concerns of eavesdropping on telephone calls or making unauthorized toll calls, but is equivalent to the security risk of the data network that connects to the APs. Different security options are supported on SpectraLink Wireless Telephones. Determining the proper level of security should be based on identified risks, corporate policy and an understanding of the pros and cons of the available security methods. 5.1 Wired Equivalent Privacy (WEP) SpectraLink Wireless Telephones support Wired Equivalent Privacy (WEP) encryption as defined by the standard. The handsets can use either 40-bit or 128-bit key lengths. WEP is intended to provide the same level of security over a wireless LAN as on a wired Ethernet LAN. Although security flaws have been identified, WEP still provides strong encryption that requires an experienced and dedicated hacker to break. While WEP is often not an acceptable option for many high security or privacy focused enterprises, it is still useful and provides reasonable performance for voice due to the shortened (quicker) key exchange process. 5.2 Wi-Fi Protected Access (WPA) Personal, WPA2 Personal Recognizing the need for stronger security standards beyond WEP, the IEEE developed the i standard, which includes stronger encryption, key management, and authentication mechanisms. Wi-Fi Protected Access (WPA) is based on draft 3.0 of the i specification and uses TKIP (Temporal Key Integrity Protocol) encryption. WPA2 is based on the ratified i standard. The major enhancement of WPA2 over WPA is the inclusion of the Advanced Encryption Standard (AES), which is widely accepted as one of the most secure encryption algorithms available. Personal mode uses a password-based authentication method called Pre-Shared Key (PSK). Personal mode is good for time-sensitive applications such as voice, because the key exchange sequence is limited and does not adversely affect roaming between APs. The PSK can be entered in hexadecimal or as an ASCII passphrase from the handset s administration menu or the HAT. The handset supports both WPA Personal and WPA2 Personal modes Cisco Fast Secure Roaming (FSR) Cisco s Fast Secure Roaming (FSR) mechanism uses a combination of standards-based and proprietary security components including Cisco Client Key Management (CCKM), LEAP authentication, Michael message integrity check (MIC) and Temporal Key Integrity Protocol (TKIP). FSR provides strong security measures for authentication, privacy and data integrity along with fast AP roaming on Cisco APs. 5.3 Using Virtual LANs Virtual LANs (VLANs) can be used to segregate traffic into different security classes. By using separate VLANs, data traffic can utilize the most robust but processing-intensive wireless security methods. In order for voice to operate efficiently in a WLAN, it is critical that it be separated from the data traffic by using VLANs, mapped to WLAN SSIDs. The 802.1Q standard establishes a method for inserting VLAN membership information into Ethernet frames via header-information tags. SpectraLink infrastructure equipment and SVP do not generate or forward these tags, but are otherwise compatible with 802.1Q up to the Ethernet switch ports used for the SpectraLink equipment. 5.4 MAC Filtering and Authentication Most access points can be configured to allow or deny association of wireless clients based on their unique MAC address, which can be used as a method of securing the WLAN. This process generally works well, but can cause some performance issues on some APs and is never recommended when using voice on a WLAN. 23

24 5.5 Firewalls and Traffic Filtering The traffic filtering capabilities of firewalls, Ethernet switches and wireless controllers can also be used as an additional security layer if configured to allow only certain types of traffic to pass onto specific areas of the LAN. To properly provide access control, it is necessary to understand the type of IP traffic used by the SpectraLink handsets. When using SpectraLink Telephony Gateways to interface to a traditional PBX or an SVP Server in an IP PBX implementation, the handset uses the SpectraLink Radio IP Protocol (ID 119). While the SpectraLink handset will generally work through a firewall if the appropriate ports are made available, this is never recommended. Firewalls create a great deal of jitter in the network which can severely limit the successful, on-time delivery of audio packets to the wireless telephone. Additionally, the use of ICMP redirects is not supported because of the extreme delay this can result when the network gateway of the SVP Server or handsets is changed dynamically. SpectraLink handset requires less than one millisecond of jitter from the SVP Server to handset. This will be difficult to achieve if there are multiple hops between the SVP Server and the handset. For an IP telephony server interface, the ports used depend on the IP telephony protocol of the telephony switch interface. The SpectraLink Wireless Telephones, Telephony Gateways and SVP Server use TCP and UDP and other common IP protocols from time to time. These include DHCP, DNS, WINS, TFTP, FTP, NTP, Telnet, ARP and ICMP. Polycom uses proprietary UDP channels between the infrastructure components i.e. UDP ports The push-to-talk (PTT) mode of the SpectraLink i640 Wireless Telephone uses the multicast IP address , which other model handsets and SpectraLink infrastructure components also employ to locate and maintain connection with each other. Some other common ports between the SVP Server and call server will be RTP traffic on ports through The port used will be chosen randomly by the phone and call server at the time of call setup. 5.6 Virtual Private Networks (VPNs) Virtual Private Networks (VPNs) are secure, private network connections. VPNs typically employ some combination of strong encryption, digital certificates, strong user authentication and access control to provide maximum security to the traffic they carry. They usually provide connectivity to many devices behind a VPN concentrator. The network can be broken into two portions - protected and unprotected: 1) The area behind the VPN server is referred to as the protected portion of the network. Sensitive, private network equipment such as file servers, servers and databases reside in this portion. 2) The area in front of the VPN server is referred to as the unprotected network, where the wireless APs and less sensitive network equipment often reside. VPNs offer an extremely effective method for securing a wireless network. Many network administrators implement VPNs to maintain the integrity of their WLANs by requiring wireless users who need access to the protected portion of the network to connect through a VPN server. Most voice devices, such as the SpectraLink Wireless Telephones, do not require access to the protected portion of the network (see Figure 8). Placing the handsets, SVP Server(s) and Telephony Gateways on the unprotected network and requiring data users to connect to the VPN ensures that the network is protected against hackers seeking to access sensitive information within the network core. 24

25 Figure 8 - Deploying SpectraLink Wireless Telephones with a VPN 5.7 Diagnostic Tools The SpectraLink handset provides three comprehensive diagnostic tools to assist the administrator in evaluating the functionality of the handsets and the surrounding wireless infrastructure. These tools are: Run Site Survey, Diagnostics Enabled, and Syslog Mode. Site Survey can be used to evaluate the coverage within the facility where the handsets are deployed by testing the signal strength, or to gather information about access points regardless of the SSID. Diagnostics Enabled is used to evaluate the overall quality of the link between the handset, access point, and other infrastructure equipment such as IP PBX, SVP Server, and gateways. The handset s diagnostics are enabled through the handset admin menu and are used while the handset is in the off hook state. Syslog Mode allows the handset to send various Syslog messages such as Successful and Failed handoffs along with reason codes indicating why the handset chose to handoff to a particular AP; Call Starts and Ends; audio statistics; and security errors. Refer to the Diagnostic Tools section of the SpectraLink e340/h340/i640 Wireless Telephone Administration Guide for a detailed explanation of information provided by each of the Diagnostic tools. 25

26 6 Subnets, Network Performance and DHCP Subnets are used to create a boundary between network segments. Although these boundaries are logical, they become like a physical boundary for mobile network devices moving throughout the enterprise. When a device with an established IP data stream (such as with an active phone call) attempts to roam across a subnet boundary, it must obtain a valid IP address within the new subnet. During this process, the data stream cannot be re-established automatically and the connection (voice call) is dropped. In the case of SpectraLink Wireless Telephones, the handsets should be power cycled to obtain a new DHCP IP address. The handsets can automatically recover in the new subnet from a lost network connection with the original subnet, but the 40-second failure and recovery time generally warrants cycling the power. Please note that in order for the phone to continue functioning in the new subnet the DHCP scope must contain the appropriate DHCP options to allow the phone to regain connectivity with the voice infrastructure. Some APs, Ethernet switches and third-party devices have implemented methods to facilitate subnet roaming. While these methods are transparent to the client device and are fundamentally a good approach to accommodating multiple subnets, they often cause enough delay and jitter to manifest poor voice quality and the tradeoffs might make such solutions unattractive for voice applications. Since the push-to-talk feature of the SpectraLink i640 Wireless Telephones use multicast IP packets, a PTT call will generally be isolated to a single subnet. With the deployment of IP multicast routing it is possible for the multicast traffic that is normally pruned at the network boundary to be passed into one or more other subnets. Please review your network manufacturer s documentation for information on how to properly configure multicast routing. There are additional subnet requirements for Wireless Telephones based on the infrastructure components that are used, as described in the following sections. 6.1 Subnets and Telephony Gateway Interfaces SpectraLink Wireless Telephones, Telephony Gateways, SVP Server(s) and the APs generally must reside on the same subnet. This is required because SpectraLink handsets use IP multicast messages to initialize the handset registration on the Telephony Gateways. In addition, The Telephony Gateways and SVP Server(s) use multicast to discover each other and stay synchronized. Most routers deployed in multi-subnet Ethernet environments are configured to filter out multicast and broadcast messages. Unless a router is configured for multicast routing, if a handset is powered up on a different subnet than the Telephony Gateway to which it is registered, the multicast message will not reach the Telephony Gateway. 6.2 Subnets and IP Telephony Server Interfaces With an IP telephony interface, the SVP Server can be placed on a separate subnet from either the APs or call server. The handsets will find the SVP Server and call server on another subnet through the default gateway option statically configured in the handset or via DHCP option 3 when using a DHCP server for IP addressing. SpectraLink Wireless Telephones can be deployed across multiple subnets when used with an IP telephony server if the performance requirements outlined below are met. One of two deployment scenarios described in this section can be used, depending on needs and infrastructure capabilities. Keep in mind that the handsets will never actively roam across a subnet boundary without power-cycling the handsets unless a VIEW Certified layer-3 roaming infrastructure is used in accordance with the VIEW deployment guidelines. In one deployment scenario for accommodating multiple subnets, each subnet is treated independently with respect to the SVP Servers and wireless network, but each subnet can still provide service to a single IP telephony server. One or more SVP Server(s) can be deployed on each subnet just as with a single subnet system, including identifying the registration SVP via DHCP option 151 or static configuration. In the second scenario, a single SVP Server (or set of SVP Servers with one registration SVP) is deployed, generally on the same subnet as the IP telephony server. The single (Registration) SVP Server is identified to all phones via DHCP option 151 or static 26

27 configuration, regardless of what subnet the phone is operating in. This scenario requires fewer SVP Servers to be installed, but requires higher performance from the router (see performance requirements in Section 6.3) The ability to cross a subnet boundary exists in either scenario, but the SpectraLink handsets will need to be power cycled to obtain a new IP address within the new subnet. In addition, other configuration considerations must be addressed. Because users will not want to re-administer the wireless telephones to a separate subnet, Extended Service Set Identifier (ESSIDs) should be the same or the handsets should be set to the Learn Always mode, the security mode and associated key should be the same or turned off, and DHCP should be used. 6.3 Network Performance Requirements Ethernet packets containing voice as their payload have short, useful lifetimes, making the timely delivery of voice packets essential. Routers can introduce latency and delay between the SVP Server and the APs, resulting in poor voice quality. Ethernet connectivity from the call server or other voice endpoint to the SVP Server should never exceed 100 milliseconds of network delay (one way), 30 milliseconds of network jitter, and 2 percent packet loss end-to-end, regardless of the physical properties of the link. The link from the SVP Server to the APs should be under 100 milliseconds of network delay, one millisecond of jitter and less than two percent packet loss. In both cases, the jitter requirements are for wired network jitter and do not include the RF link. One function of the SVP Server is to control the timing of packets through the AP. The SVP server delivers audio packets to the wireless telephone every 30 millisecond. The delay between the SVP Server and the AP needs to be controlled and consistent. Wired QoS (DSCP) is one aspect of ensuring voice packets have highest priority. The jitter requirement between the call server and the SVP Server is a function of how the audio is packetized for encapsulation in the SpectraLink Radio Protocol (SRP) and the packet queuing in the SVP Server. Jitter between the SVP Server and the AP should be measured at the wired Ethernet connection to the AP. If the AP is a lightweight AP attached to a wireless controller and Polycom has VIEW Certified the system, jitter can be measured at the entry to the wireless controller. However it is better to measure jitter at the AP s Ethernet interface if the AP does not connect directly to the wireless controller. For this measurement, the SVP Server is delivering packets at 30 millisecond intervals with no jitter. The time is measured from the arrival of one packet from the SVP Server directed to a single wireless telephone to the next packet from the SVP Server to the same wireless telephone. The jitter measurement is the time difference from the ideal 30 millisecond arrival of packets at the AP. See Figure 9. 27

28 Figure 9 - Measuring Network Jitter In a multiple SVP Server configuration, jitter is measured from the SVP Server that is responsible for the traffic through a given AP to a wireless telephone. This may be different than the SVP Server that is acting as a proxy for the wireless telephone to the IP PBX. Refer to Section for additional multiple SVP Server information. SpectraLink handsets have a diagnostic option that includes jitter measurement. The calculated jitter shown in this mode is not the jitter described above because it includes delays in the AP, radio link and queue times inside the wireless telephone. Jitter information from the handset diagnostic mode should only be used as a guideline for diagnosing major network or radio link problems. 6.4 DHCP Requirements The SpectraLink Wireless Telephone is configured by default to utilize DHCP in order to obtain an IP address and the necessary DHCP options to allow the phone to operate normally. For the different IP protocols available with the SpectraLink handset there are a number of different DHCP options that would be used. However, there are also a number of DHCP options that are universal required regardless of the IP protocol implementation. Additionally, the DHCP options required for a Telephony Gateway implementation vary somewhat from the IP protocol deployments. Operationally, the handset functions the same regardless of the implementation when using DHCP. This means all that is changing is the DHCP options the handset will require in order to function properly. For Telephony Gateway deployments the handset will acquire its IP address from the DHCP server Scope Address Pool and its Subnet mask in addition to the most basic options from the DHCP server Scope Options. The expectation when deploying a Telephony Gateway system is that the handset will reside on the same IP subnet as the Gateway in order to generate the necessary multicast frames that will allow it to locate and register with the Telephony Gateway. Once the handset registers, the remaining options that would normally be handled via DHCP, such as TFTP, are provided by the Telephony Gateway. When the license option in the wireless phone is set for one of the IP protocols the handset will require several other DHCP options in order to function properly. In addition to the base DHCP options required by the Telephony 28

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